FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/speexdec.c
Date: 2022-10-02 18:56:10
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1 /*
2 * Copyright 2002-2008 Xiph.org Foundation
3 * Copyright 2002-2008 Jean-Marc Valin
4 * Copyright 2005-2007 Analog Devices Inc.
5 * Copyright 2005-2008 Commonwealth Scientific and Industrial Research Organisation (CSIRO)
6 * Copyright 1993, 2002, 2006 David Rowe
7 * Copyright 2003 EpicGames
8 * Copyright 1992-1994 Jutta Degener, Carsten Bormann
9
10 * Redistribution and use in source and binary forms, with or without
11 * modification, are permitted provided that the following conditions
12 * are met:
13
14 * - Redistributions of source code must retain the above copyright
15 * notice, this list of conditions and the following disclaimer.
16
17 * - Redistributions in binary form must reproduce the above copyright
18 * notice, this list of conditions and the following disclaimer in the
19 * documentation and/or other materials provided with the distribution.
20
21 * - Neither the name of the Xiph.org Foundation nor the names of its
22 * contributors may be used to endorse or promote products derived from
23 * this software without specific prior written permission.
24
25 * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
26 * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
27 * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
28 * A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
29 * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
30 * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
31 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
32 * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
33 * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
34 * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
35 * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
36 *
37 * This file is part of FFmpeg.
38 *
39 * FFmpeg is free software; you can redistribute it and/or
40 * modify it under the terms of the GNU Lesser General Public
41 * License as published by the Free Software Foundation; either
42 * version 2.1 of the License, or (at your option) any later version.
43 *
44 * FFmpeg is distributed in the hope that it will be useful,
45 * but WITHOUT ANY WARRANTY; without even the implied warranty of
46 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
47 * Lesser General Public License for more details.
48 *
49 * You should have received a copy of the GNU Lesser General Public
50 * License along with FFmpeg; if not, write to the Free Software
51 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
52 */
53
54 #include "libavutil/avassert.h"
55 #include "libavutil/float_dsp.h"
56 #include "avcodec.h"
57 #include "bytestream.h"
58 #include "codec_internal.h"
59 #include "decode.h"
60 #include "get_bits.h"
61 #include "speexdata.h"
62
63 #define SPEEX_NB_MODES 3
64 #define SPEEX_INBAND_STEREO 9
65
66 #define QMF_ORDER 64
67 #define NB_ORDER 10
68 #define NB_FRAME_SIZE 160
69 #define NB_SUBMODES 9
70 #define NB_SUBMODE_BITS 4
71 #define SB_SUBMODE_BITS 3
72
73 #define NB_SUBFRAME_SIZE 40
74 #define NB_NB_SUBFRAMES 4
75 #define NB_PITCH_START 17
76 #define NB_PITCH_END 144
77
78 #define NB_DEC_BUFFER (NB_FRAME_SIZE + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12)
79
80 #define SPEEX_MEMSET(dst, c, n) (memset((dst), (c), (n) * sizeof(*(dst))))
81 #define SPEEX_COPY(dst, src, n) (memcpy((dst), (src), (n) * sizeof(*(dst))))
82
83 #define LSP_LINEAR(i) (.25f * (i) + .25f)
84 #define LSP_LINEAR_HIGH(i) (.3125f * (i) + .75f)
85 #define LSP_DIV_256(x) (0.00390625f * (x))
86 #define LSP_DIV_512(x) (0.001953125f * (x))
87 #define LSP_DIV_1024(x) (0.0009765625f * (x))
88
89 typedef struct LtpParams {
90 const int8_t *gain_cdbk;
91 int gain_bits;
92 int pitch_bits;
93 } LtpParam;
94
95 static const LtpParam ltp_params_vlbr = { gain_cdbk_lbr, 5, 0 };
96 static const LtpParam ltp_params_lbr = { gain_cdbk_lbr, 5, 7 };
97 static const LtpParam ltp_params_med = { gain_cdbk_lbr, 5, 7 };
98 static const LtpParam ltp_params_nb = { gain_cdbk_nb, 7, 7 };
99
100 typedef struct SplitCodebookParams {
101 int subvect_size;
102 int nb_subvect;
103 const signed char *shape_cb;
104 int shape_bits;
105 int have_sign;
106 } SplitCodebookParams;
107
108 static const SplitCodebookParams split_cb_nb_ulbr = { 20, 2, exc_20_32_table, 5, 0 };
109 static const SplitCodebookParams split_cb_nb_vlbr = { 10, 4, exc_10_16_table, 4, 0 };
110 static const SplitCodebookParams split_cb_nb_lbr = { 10, 4, exc_10_32_table, 5, 0 };
111 static const SplitCodebookParams split_cb_nb_med = { 8, 5, exc_8_128_table, 7, 0 };
112 static const SplitCodebookParams split_cb_nb = { 5, 8, exc_5_64_table, 6, 0 };
113 static const SplitCodebookParams split_cb_sb = { 5, 8, exc_5_256_table, 8, 0 };
114 static const SplitCodebookParams split_cb_high = { 8, 5, hexc_table, 7, 1 };
115 static const SplitCodebookParams split_cb_high_lbr= { 10, 4, hexc_10_32_table,5, 0 };
116
117 /** Quantizes LSPs */
118 typedef void (*lsp_quant_func)(float *, float *, int, GetBitContext *);
119
120 /** Decodes quantized LSPs */
121 typedef void (*lsp_unquant_func)(float *, int, GetBitContext *);
122
123 /** Long-term predictor quantization */
124 typedef int (*ltp_quant_func)(float *, float *, float *,
125 float *, float *, float *,
126 const void *, int, int, float, int, int,
127 GetBitContext *, char *, float *,
128 float *, int, int, int, float *);
129
130 /** Long-term un-quantize */
131 typedef void (*ltp_unquant_func)(float *, float *, int, int,
132 float, const void *, int, int *,
133 float *, GetBitContext *, int, int,
134 float, int);
135
136 /** Innovation quantization function */
137 typedef void (*innovation_quant_func)(float *, float *,
138 float *, float *, const void *,
139 int, int, float *, float *,
140 GetBitContext *, char *, int, int);
141
142 /** Innovation unquantization function */
143 typedef void (*innovation_unquant_func)(float *, const void *, int,
144 GetBitContext *, uint32_t *);
145
146 typedef struct SpeexSubmode {
147 int lbr_pitch; /**< Set to -1 for "normal" modes, otherwise encode pitch using
148 a global pitch and allowing a +- lbr_pitch variation (for
149 low not-rates)*/
150 int forced_pitch_gain; /**< Use the same (forced) pitch gain for all
151 sub-frames */
152 int have_subframe_gain; /**< Number of bits to use as sub-frame innovation
153 gain */
154 int double_codebook; /**< Apply innovation quantization twice for higher
155 quality (and higher bit-rate)*/
156 lsp_unquant_func lsp_unquant; /**< LSP unquantization function */
157
158 ltp_unquant_func ltp_unquant; /**< Long-term predictor (pitch) un-quantizer */
159 const void *LtpParam; /**< Pitch parameters (options) */
160
161 innovation_unquant_func innovation_unquant; /**< Innovation un-quantization */
162 const void *innovation_params; /**< Innovation quantization parameters*/
163
164 float comb_gain; /**< Gain of enhancer comb filter */
165 } SpeexSubmode;
166
167 typedef struct SpeexMode {
168 int modeID; /**< ID of the mode */
169 int (*decode)(AVCodecContext *avctx, void *dec, GetBitContext *gb, float *out);
170 int frame_size; /**< Size of frames used for decoding */
171 int subframe_size; /**< Size of sub-frames used for decoding */
172 int lpc_size; /**< Order of LPC filter */
173 float folding_gain; /**< Folding gain */
174 const SpeexSubmode *submodes[NB_SUBMODES]; /**< Sub-mode data for the mode */
175 int default_submode; /**< Default sub-mode to use when decoding */
176 } SpeexMode;
177
178 typedef struct DecoderState {
179 const SpeexMode *mode;
180 int modeID; /**< ID of the decoder mode */
181 int first; /**< Is first frame */
182 int full_frame_size; /**< Length of full-band frames */
183 int is_wideband; /**< If wideband is present */
184 int count_lost; /**< Was the last frame lost? */
185 int frame_size; /**< Length of high-band frames */
186 int subframe_size; /**< Length of high-band sub-frames */
187 int nb_subframes; /**< Number of high-band sub-frames */
188 int lpc_size; /**< Order of high-band LPC analysis */
189 float last_ol_gain; /**< Open-loop gain for previous frame */
190 float *innov_save; /**< If non-NULL, innovation is copied here */
191
192 /* This is used in packet loss concealment */
193 int last_pitch; /**< Pitch of last correctly decoded frame */
194 float last_pitch_gain; /**< Pitch gain of last correctly decoded frame */
195 uint32_t seed; /**< Seed used for random number generation */
196
197 int encode_submode;
198 const SpeexSubmode *const *submodes; /**< Sub-mode data */
199 int submodeID; /**< Activated sub-mode */
200 int lpc_enh_enabled; /**< 1 when LPC enhancer is on, 0 otherwise */
201
202 /* Vocoder data */
203 float voc_m1;
204 float voc_m2;
205 float voc_mean;
206 int voc_offset;
207
208 int dtx_enabled;
209 int highpass_enabled; /**< Is the input filter enabled */
210
211 float *exc; /**< Start of excitation frame */
212 float mem_hp[2]; /**< High-pass filter memory */
213 float exc_buf[NB_DEC_BUFFER]; /**< Excitation buffer */
214 float old_qlsp[NB_ORDER]; /**< Quantized LSPs for previous frame */
215 float interp_qlpc[NB_ORDER]; /**< Interpolated quantized LPCs */
216 float mem_sp[NB_ORDER]; /**< Filter memory for synthesis signal */
217 float g0_mem[QMF_ORDER];
218 float g1_mem[QMF_ORDER];
219 float pi_gain[NB_NB_SUBFRAMES]; /**< Gain of LPC filter at theta=pi (fe/2) */
220 float exc_rms[NB_NB_SUBFRAMES]; /**< RMS of excitation per subframe */
221 } DecoderState;
222
223 /* Default handler for user callbacks: skip it */
224 static int speex_default_user_handler(GetBitContext *gb, void *state, void *data)
225 {
226 const int req_size = get_bits(gb, 4);
227 skip_bits_long(gb, 5 + 8 * req_size);
228 return 0;
229 }
230
231 typedef struct StereoState {
232 float balance; /**< Left/right balance info */
233 float e_ratio; /**< Ratio of energies: E(left+right)/[E(left)+E(right)] */
234 float smooth_left; /**< Smoothed left channel gain */
235 float smooth_right; /**< Smoothed right channel gain */
236 } StereoState;
237
238 typedef struct SpeexContext {
239 AVClass *class;
240 GetBitContext gb;
241
242 int32_t version_id; /**< Version for Speex (for checking compatibility) */
243 int32_t rate; /**< Sampling rate used */
244 int32_t mode; /**< Mode used (0 for narrowband, 1 for wideband) */
245 int32_t bitstream_version; /**< Version ID of the bit-stream */
246 int32_t nb_channels; /**< Number of channels decoded */
247 int32_t bitrate; /**< Bit-rate used */
248 int32_t frame_size; /**< Size of frames */
249 int32_t vbr; /**< 1 for a VBR decoding, 0 otherwise */
250 int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */
251 int32_t extra_headers; /**< Number of additional headers after the comments */
252
253 int pkt_size;
254
255 StereoState stereo;
256 DecoderState st[SPEEX_NB_MODES];
257
258 AVFloatDSPContext *fdsp;
259 } SpeexContext;
260
261 static void lsp_unquant_lbr(float *lsp, int order, GetBitContext *gb)
262 {
263 int id;
264
265 for (int i = 0; i < order; i++)
266 lsp[i] = LSP_LINEAR(i);
267
268 id = get_bits(gb, 6);
269 for (int i = 0; i < 10; i++)
270 lsp[i] += LSP_DIV_256(cdbk_nb[id * 10 + i]);
271
272 id = get_bits(gb, 6);
273 for (int i = 0; i < 5; i++)
274 lsp[i] += LSP_DIV_512(cdbk_nb_low1[id * 5 + i]);
275
276 id = get_bits(gb, 6);
277 for (int i = 0; i < 5; i++)
278 lsp[i + 5] += LSP_DIV_512(cdbk_nb_high1[id * 5 + i]);
279 }
280
281 static void forced_pitch_unquant(float *exc, float *exc_out, int start, int end,
282 float pitch_coef, const void *par, int nsf,
283 int *pitch_val, float *gain_val, GetBitContext *gb, int count_lost,
284 int subframe_offset, float last_pitch_gain, int cdbk_offset)
285 {
286 av_assert0(!isnan(pitch_coef));
287 pitch_coef = fminf(pitch_coef, .99f);
288 for (int i = 0; i < nsf; i++) {
289 exc_out[i] = exc[i - start] * pitch_coef;
290 exc[i] = exc_out[i];
291 }
292 pitch_val[0] = start;
293 gain_val[0] = gain_val[2] = 0.f;
294 gain_val[1] = pitch_coef;
295 }
296
297 static inline float speex_rand(float std, uint32_t *seed)
298 {
299 const uint32_t jflone = 0x3f800000;
300 const uint32_t jflmsk = 0x007fffff;
301 float fran;
302 uint32_t ran;
303 seed[0] = 1664525 * seed[0] + 1013904223;
304 ran = jflone | (jflmsk & seed[0]);
305 fran = av_int2float(ran);
306 fran -= 1.5f;
307 fran *= std;
308 return fran;
309 }
310
311 static void noise_codebook_unquant(float *exc, const void *par, int nsf,
312 GetBitContext *gb, uint32_t *seed)
313 {
314 for (int i = 0; i < nsf; i++)
315 exc[i] = speex_rand(1.f, seed);
316 }
317
318 static void split_cb_shape_sign_unquant(float *exc, const void *par, int nsf,
319 GetBitContext *gb, uint32_t *seed)
320 {
321 int subvect_size, nb_subvect, have_sign, shape_bits;
322 const SplitCodebookParams *params;
323 const signed char *shape_cb;
324 int signs[10], ind[10];
325
326 params = par;
327 subvect_size = params->subvect_size;
328 nb_subvect = params->nb_subvect;
329
330 shape_cb = params->shape_cb;
331 have_sign = params->have_sign;
332 shape_bits = params->shape_bits;
333
334 /* Decode codewords and gains */
335 for (int i = 0; i < nb_subvect; i++) {
336 signs[i] = have_sign ? get_bits1(gb) : 0;
337 ind[i] = get_bitsz(gb, shape_bits);
338 }
339 /* Compute decoded excitation */
340 for (int i = 0; i < nb_subvect; i++) {
341 const float s = signs[i] ? -1.f : 1.f;
342
343 for (int j = 0; j < subvect_size; j++)
344 exc[subvect_size * i + j] += s * 0.03125f * shape_cb[ind[i] * subvect_size + j];
345 }
346 }
347
348 #define SUBMODE(x) st->submodes[st->submodeID]->x
349
350 #define gain_3tap_to_1tap(g) (FFABS(g[1]) + (g[0] > 0.f ? g[0] : -.5f * g[0]) + (g[2] > 0.f ? g[2] : -.5f * g[2]))
351
352 static void
353 pitch_unquant_3tap(float *exc, float *exc_out, int start, int end, float pitch_coef,
354 const void *par, int nsf, int *pitch_val, float *gain_val, GetBitContext *gb,
355 int count_lost, int subframe_offset, float last_pitch_gain, int cdbk_offset)
356 {
357 int pitch, gain_index, gain_cdbk_size;
358 const int8_t *gain_cdbk;
359 const LtpParam *params;
360 float gain[3];
361
362 params = (const LtpParam *)par;
363 gain_cdbk_size = 1 << params->gain_bits;
364 gain_cdbk = params->gain_cdbk + 4 * gain_cdbk_size * cdbk_offset;
365
366 pitch = get_bitsz(gb, params->pitch_bits);
367 pitch += start;
368 gain_index = get_bitsz(gb, params->gain_bits);
369 gain[0] = 0.015625f * gain_cdbk[gain_index * 4] + .5f;
370 gain[1] = 0.015625f * gain_cdbk[gain_index * 4 + 1] + .5f;
371 gain[2] = 0.015625f * gain_cdbk[gain_index * 4 + 2] + .5f;
372
373 if (count_lost && pitch > subframe_offset) {
374 float tmp = count_lost < 4 ? last_pitch_gain : 0.5f * last_pitch_gain;
375 float gain_sum;
376
377 tmp = fminf(tmp, .95f);
378 gain_sum = gain_3tap_to_1tap(gain);
379
380 if (gain_sum > tmp && gain_sum > 0.f) {
381 float fact = tmp / gain_sum;
382 for (int i = 0; i < 3; i++)
383 gain[i] *= fact;
384 }
385 }
386
387 pitch_val[0] = pitch;
388 gain_val[0] = gain[0];
389 gain_val[1] = gain[1];
390 gain_val[2] = gain[2];
391 SPEEX_MEMSET(exc_out, 0, nsf);
392
393 for (int i = 0; i < 3; i++) {
394 int tmp1, tmp3;
395 int pp = pitch + 1 - i;
396 tmp1 = nsf;
397 if (tmp1 > pp)
398 tmp1 = pp;
399 for (int j = 0; j < tmp1; j++)
400 exc_out[j] += gain[2 - i] * exc[j - pp];
401 tmp3 = nsf;
402 if (tmp3 > pp + pitch)
403 tmp3 = pp + pitch;
404 for (int j = tmp1; j < tmp3; j++)
405 exc_out[j] += gain[2 - i] * exc[j - pp - pitch];
406 }
407 }
408
409 static void lsp_unquant_nb(float *lsp, int order, GetBitContext *gb)
410 {
411 int id;
412
413 for (int i = 0; i < order; i++)
414 lsp[i] = LSP_LINEAR(i);
415
416 id = get_bits(gb, 6);
417 for (int i = 0; i < 10; i++)
418 lsp[i] += LSP_DIV_256(cdbk_nb[id * 10 + i]);
419
420 id = get_bits(gb, 6);
421 for (int i = 0; i < 5; i++)
422 lsp[i] += LSP_DIV_512(cdbk_nb_low1[id * 5 + i]);
423
424 id = get_bits(gb, 6);
425 for (int i = 0; i < 5; i++)
426 lsp[i] += LSP_DIV_1024(cdbk_nb_low2[id * 5 + i]);
427
428 id = get_bits(gb, 6);
429 for (int i = 0; i < 5; i++)
430 lsp[i + 5] += LSP_DIV_512(cdbk_nb_high1[id * 5 + i]);
431
432 id = get_bits(gb, 6);
433 for (int i = 0; i < 5; i++)
434 lsp[i + 5] += LSP_DIV_1024(cdbk_nb_high2[id * 5 + i]);
435 }
436
437 static void lsp_unquant_high(float *lsp, int order, GetBitContext *gb)
438 {
439 int id;
440
441 for (int i = 0; i < order; i++)
442 lsp[i] = LSP_LINEAR_HIGH(i);
443
444 id = get_bits(gb, 6);
445 for (int i = 0; i < order; i++)
446 lsp[i] += LSP_DIV_256(high_lsp_cdbk[id * order + i]);
447
448 id = get_bits(gb, 6);
449 for (int i = 0; i < order; i++)
450 lsp[i] += LSP_DIV_512(high_lsp_cdbk2[id * order + i]);
451 }
452
453 /* 2150 bps "vocoder-like" mode for comfort noise */
454 static const SpeexSubmode nb_submode1 = {
455 0, 1, 0, 0, lsp_unquant_lbr, forced_pitch_unquant, NULL,
456 noise_codebook_unquant, NULL, -1.f
457 };
458
459 /* 5.95 kbps very low bit-rate mode */
460 static const SpeexSubmode nb_submode2 = {
461 0, 0, 0, 0, lsp_unquant_lbr, pitch_unquant_3tap, &ltp_params_vlbr,
462 split_cb_shape_sign_unquant, &split_cb_nb_vlbr, .6f
463 };
464
465 /* 8 kbps low bit-rate mode */
466 static const SpeexSubmode nb_submode3 = {
467 -1, 0, 1, 0, lsp_unquant_lbr, pitch_unquant_3tap, &ltp_params_lbr,
468 split_cb_shape_sign_unquant, &split_cb_nb_lbr, .55f
469 };
470
471 /* 11 kbps medium bit-rate mode */
472 static const SpeexSubmode nb_submode4 = {
473 -1, 0, 1, 0, lsp_unquant_lbr, pitch_unquant_3tap, &ltp_params_med,
474 split_cb_shape_sign_unquant, &split_cb_nb_med, .45f
475 };
476
477 /* 15 kbps high bit-rate mode */
478 static const SpeexSubmode nb_submode5 = {
479 -1, 0, 3, 0, lsp_unquant_nb, pitch_unquant_3tap, &ltp_params_nb,
480 split_cb_shape_sign_unquant, &split_cb_nb, .25f
481 };
482
483 /* 18.2 high bit-rate mode */
484 static const SpeexSubmode nb_submode6 = {
485 -1, 0, 3, 0, lsp_unquant_nb, pitch_unquant_3tap, &ltp_params_nb,
486 split_cb_shape_sign_unquant, &split_cb_sb, .15f
487 };
488
489 /* 24.6 kbps high bit-rate mode */
490 static const SpeexSubmode nb_submode7 = {
491 -1, 0, 3, 1, lsp_unquant_nb, pitch_unquant_3tap, &ltp_params_nb,
492 split_cb_shape_sign_unquant, &split_cb_nb, 0.05f
493 };
494
495 /* 3.95 kbps very low bit-rate mode */
496 static const SpeexSubmode nb_submode8 = {
497 0, 1, 0, 0, lsp_unquant_lbr, forced_pitch_unquant, NULL,
498 split_cb_shape_sign_unquant, &split_cb_nb_ulbr, .5f
499 };
500
501 static const SpeexSubmode wb_submode1 = {
502 0, 0, 1, 0, lsp_unquant_high, NULL, NULL,
503 NULL, NULL, -1.f
504 };
505
506 static const SpeexSubmode wb_submode2 = {
507 0, 0, 1, 0, lsp_unquant_high, NULL, NULL,
508 split_cb_shape_sign_unquant, &split_cb_high_lbr, -1.f
509 };
510
511 static const SpeexSubmode wb_submode3 = {
512 0, 0, 1, 0, lsp_unquant_high, NULL, NULL,
513 split_cb_shape_sign_unquant, &split_cb_high, -1.f
514 };
515
516 static const SpeexSubmode wb_submode4 = {
517 0, 0, 1, 1, lsp_unquant_high, NULL, NULL,
518 split_cb_shape_sign_unquant, &split_cb_high, -1.f
519 };
520
521 static int nb_decode(AVCodecContext *, void *, GetBitContext *, float *);
522 static int sb_decode(AVCodecContext *, void *, GetBitContext *, float *);
523
524 static const SpeexMode speex_modes[SPEEX_NB_MODES] = {
525 {
526 .modeID = 0,
527 .decode = nb_decode,
528 .frame_size = NB_FRAME_SIZE,
529 .subframe_size = NB_SUBFRAME_SIZE,
530 .lpc_size = NB_ORDER,
531 .submodes = {
532 NULL, &nb_submode1, &nb_submode2, &nb_submode3, &nb_submode4,
533 &nb_submode5, &nb_submode6, &nb_submode7, &nb_submode8
534 },
535 .default_submode = 5,
536 },
537 {
538 .modeID = 1,
539 .decode = sb_decode,
540 .frame_size = NB_FRAME_SIZE,
541 .subframe_size = NB_SUBFRAME_SIZE,
542 .lpc_size = 8,
543 .folding_gain = 0.9f,
544 .submodes = {
545 NULL, &wb_submode1, &wb_submode2, &wb_submode3, &wb_submode4
546 },
547 .default_submode = 3,
548 },
549 {
550 .modeID = 2,
551 .decode = sb_decode,
552 .frame_size = 320,
553 .subframe_size = 80,
554 .lpc_size = 8,
555 .folding_gain = 0.7f,
556 .submodes = {
557 NULL, &wb_submode1
558 },
559 .default_submode = 1,
560 },
561 };
562
563 static float compute_rms(const float *x, int len)
564 {
565 float sum = 0.f;
566
567 for (int i = 0; i < len; i++)
568 sum += x[i] * x[i];
569
570 av_assert0(len > 0);
571 return sqrtf(.1f + sum / len);
572 }
573
574 static void bw_lpc(float gamma, const float *lpc_in,
575 float *lpc_out, int order)
576 {
577 float tmp = gamma;
578
579 for (int i = 0; i < order; i++) {
580 lpc_out[i] = tmp * lpc_in[i];
581 tmp *= gamma;
582 }
583 }
584
585 static void iir_mem(const float *x, const float *den,
586 float *y, int N, int ord, float *mem)
587 {
588 for (int i = 0; i < N; i++) {
589 float yi = x[i] + mem[0];
590 float nyi = -yi;
591 for (int j = 0; j < ord - 1; j++)
592 mem[j] = mem[j + 1] + den[j] * nyi;
593 mem[ord - 1] = den[ord - 1] * nyi;
594 y[i] = yi;
595 }
596 }
597
598 static void highpass(const float *x, float *y, int len, float *mem, int wide)
599 {
600 static const float Pcoef[2][3] = {{ 1.00000f, -1.92683f, 0.93071f }, { 1.00000f, -1.97226f, 0.97332f } };
601 static const float Zcoef[2][3] = {{ 0.96446f, -1.92879f, 0.96446f }, { 0.98645f, -1.97277f, 0.98645f } };
602 const float *den, *num;
603
604 den = Pcoef[wide];
605 num = Zcoef[wide];
606 for (int i = 0; i < len; i++) {
607 float yi = num[0] * x[i] + mem[0];
608 mem[0] = mem[1] + num[1] * x[i] + -den[1] * yi;
609 mem[1] = num[2] * x[i] + -den[2] * yi;
610 y[i] = yi;
611 }
612 }
613
614 #define median3(a, b, c) \
615 ((a) < (b) ? ((b) < (c) ? (b) : ((a) < (c) ? (c) : (a))) \
616 : ((c) < (b) ? (b) : ((c) < (a) ? (c) : (a))))
617
618 static int speex_std_stereo(GetBitContext *gb, void *state, void *data)
619 {
620 StereoState *stereo = data;
621 float sign = get_bits1(gb) ? -1.f : 1.f;
622
623 stereo->balance = exp(sign * .25f * get_bits(gb, 5));
624 stereo->e_ratio = e_ratio_quant[get_bits(gb, 2)];
625
626 return 0;
627 }
628
629 static int speex_inband_handler(GetBitContext *gb, void *state, StereoState *stereo)
630 {
631 int id = get_bits(gb, 4);
632
633 if (id == SPEEX_INBAND_STEREO) {
634 return speex_std_stereo(gb, state, stereo);
635 } else {
636 int adv;
637
638 if (id < 2)
639 adv = 1;
640 else if (id < 8)
641 adv = 4;
642 else if (id < 10)
643 adv = 8;
644 else if (id < 12)
645 adv = 16;
646 else if (id < 14)
647 adv = 32;
648 else
649 adv = 64;
650 skip_bits_long(gb, adv);
651 }
652 return 0;
653 }
654
655 static void sanitize_values(float *vec, float min_val, float max_val, int len)
656 {
657 for (int i = 0; i < len; i++) {
658 if (!isnormal(vec[i]) || fabsf(vec[i]) < 1e-8f)
659 vec[i] = 0.f;
660 else
661 vec[i] = av_clipf(vec[i], min_val, max_val);
662 }
663 }
664
665 static void signal_mul(const float *x, float *y, float scale, int len)
666 {
667 for (int i = 0; i < len; i++)
668 y[i] = scale * x[i];
669 }
670
671 static float inner_prod(const float *x, const float *y, int len)
672 {
673 float sum = 0.f;
674
675 for (int i = 0; i < len; i += 8) {
676 float part = 0.f;
677 part += x[i + 0] * y[i + 0];
678 part += x[i + 1] * y[i + 1];
679 part += x[i + 2] * y[i + 2];
680 part += x[i + 3] * y[i + 3];
681 part += x[i + 4] * y[i + 4];
682 part += x[i + 5] * y[i + 5];
683 part += x[i + 6] * y[i + 6];
684 part += x[i + 7] * y[i + 7];
685 sum += part;
686 }
687
688 return sum;
689 }
690
691 static int interp_pitch(const float *exc, float *interp, int pitch, int len)
692 {
693 float corr[4][7], maxcorr;
694 int maxi, maxj;
695
696 for (int i = 0; i < 7; i++)
697 corr[0][i] = inner_prod(exc, exc - pitch - 3 + i, len);
698 for (int i = 0; i < 3; i++) {
699 for (int j = 0; j < 7; j++) {
700 int i1, i2;
701 float tmp = 0.f;
702
703 i1 = 3 - j;
704 if (i1 < 0)
705 i1 = 0;
706 i2 = 10 - j;
707 if (i2 > 7)
708 i2 = 7;
709 for (int k = i1; k < i2; k++)
710 tmp += shift_filt[i][k] * corr[0][j + k - 3];
711 corr[i + 1][j] = tmp;
712 }
713 }
714 maxi = maxj = 0;
715 maxcorr = corr[0][0];
716 for (int i = 0; i < 4; i++) {
717 for (int j = 0; j < 7; j++) {
718 if (corr[i][j] > maxcorr) {
719 maxcorr = corr[i][j];
720 maxi = i;
721 maxj = j;
722 }
723 }
724 }
725 for (int i = 0; i < len; i++) {
726 float tmp = 0.f;
727 if (maxi > 0.f) {
728 for (int k = 0; k < 7; k++)
729 tmp += exc[i - (pitch - maxj + 3) + k - 3] * shift_filt[maxi - 1][k];
730 } else {
731 tmp = exc[i - (pitch - maxj + 3)];
732 }
733 interp[i] = tmp;
734 }
735 return pitch - maxj + 3;
736 }
737
738 static void multicomb(const float *exc, float *new_exc, float *ak, int p, int nsf,
739 int pitch, int max_pitch, float comb_gain)
740 {
741 float old_ener, new_ener;
742 float iexc0_mag, iexc1_mag, exc_mag;
743 float iexc[4 * NB_SUBFRAME_SIZE];
744 float corr0, corr1, gain0, gain1;
745 float pgain1, pgain2;
746 float c1, c2, g1, g2;
747 float ngain, gg1, gg2;
748 int corr_pitch = pitch;
749
750 interp_pitch(exc, iexc, corr_pitch, 80);
751 if (corr_pitch > max_pitch)
752 interp_pitch(exc, iexc + nsf, 2 * corr_pitch, 80);
753 else
754 interp_pitch(exc, iexc + nsf, -corr_pitch, 80);
755
756 iexc0_mag = sqrtf(1000.f + inner_prod(iexc, iexc, nsf));
757 iexc1_mag = sqrtf(1000.f + inner_prod(iexc + nsf, iexc + nsf, nsf));
758 exc_mag = sqrtf(1.f + inner_prod(exc, exc, nsf));
759 corr0 = inner_prod(iexc, exc, nsf);
760 corr1 = inner_prod(iexc + nsf, exc, nsf);
761 if (corr0 > iexc0_mag * exc_mag)
762 pgain1 = 1.f;
763 else
764 pgain1 = (corr0 / exc_mag) / iexc0_mag;
765 if (corr1 > iexc1_mag * exc_mag)
766 pgain2 = 1.f;
767 else
768 pgain2 = (corr1 / exc_mag) / iexc1_mag;
769 gg1 = exc_mag / iexc0_mag;
770 gg2 = exc_mag / iexc1_mag;
771 if (comb_gain > 0.f) {
772 c1 = .4f * comb_gain + .07f;
773 c2 = .5f + 1.72f * (c1 - .07f);
774 } else {
775 c1 = c2 = 0.f;
776 }
777 g1 = 1.f - c2 * pgain1 * pgain1;
778 g2 = 1.f - c2 * pgain2 * pgain2;
779 g1 = fmaxf(g1, c1);
780 g2 = fmaxf(g2, c1);
781 g1 = c1 / g1;
782 g2 = c1 / g2;
783
784 if (corr_pitch > max_pitch) {
785 gain0 = .7f * g1 * gg1;
786 gain1 = .3f * g2 * gg2;
787 } else {
788 gain0 = .6f * g1 * gg1;
789 gain1 = .6f * g2 * gg2;
790 }
791 for (int i = 0; i < nsf; i++)
792 new_exc[i] = exc[i] + (gain0 * iexc[i]) + (gain1 * iexc[i + nsf]);
793 new_ener = compute_rms(new_exc, nsf);
794 old_ener = compute_rms(exc, nsf);
795
796 old_ener = fmaxf(old_ener, 1.f);
797 new_ener = fmaxf(new_ener, 1.f);
798 old_ener = fminf(old_ener, new_ener);
799 ngain = old_ener / new_ener;
800
801 for (int i = 0; i < nsf; i++)
802 new_exc[i] *= ngain;
803 }
804
805 static void lsp_interpolate(const float *old_lsp, const float *new_lsp,
806 float *lsp, int len, int subframe,
807 int nb_subframes, float margin)
808 {
809 const float tmp = (1.f + subframe) / nb_subframes;
810
811 for (int i = 0; i < len; i++) {
812 lsp[i] = (1.f - tmp) * old_lsp[i] + tmp * new_lsp[i];
813 lsp[i] = av_clipf(lsp[i], margin, M_PI - margin);
814 }
815 for (int i = 1; i < len - 1; i++) {
816 lsp[i] = fmaxf(lsp[i], lsp[i - 1] + margin);
817 if (lsp[i] > lsp[i + 1] - margin)
818 lsp[i] = .5f * (lsp[i] + lsp[i + 1] - margin);
819 }
820 }
821
822 static void lsp_to_lpc(const float *freq, float *ak, int lpcrdr)
823 {
824 float xout1, xout2, xin1, xin2;
825 float *pw, *n0;
826 float Wp[4 * NB_ORDER + 2] = { 0 };
827 float x_freq[NB_ORDER];
828 const int m = lpcrdr >> 1;
829
830 pw = Wp;
831
832 xin1 = xin2 = 1.f;
833
834 for (int i = 0; i < lpcrdr; i++)
835 x_freq[i] = -cosf(freq[i]);
836
837 /* reconstruct P(z) and Q(z) by cascading second order
838 * polynomials in form 1 - 2xz(-1) +z(-2), where x is the
839 * LSP coefficient
840 */
841 for (int j = 0; j <= lpcrdr; j++) {
842 int i2 = 0;
843 for (int i = 0; i < m; i++, i2 += 2) {
844 n0 = pw + (i * 4);
845 xout1 = xin1 + 2.f * x_freq[i2 ] * n0[0] + n0[1];
846 xout2 = xin2 + 2.f * x_freq[i2 + 1] * n0[2] + n0[3];
847 n0[1] = n0[0];
848 n0[3] = n0[2];
849 n0[0] = xin1;
850 n0[2] = xin2;
851 xin1 = xout1;
852 xin2 = xout2;
853 }
854 xout1 = xin1 + n0[4];
855 xout2 = xin2 - n0[5];
856 if (j > 0)
857 ak[j - 1] = (xout1 + xout2) * 0.5f;
858 n0[4] = xin1;
859 n0[5] = xin2;
860
861 xin1 = 0.f;
862 xin2 = 0.f;
863 }
864 }
865
866 static int nb_decode(AVCodecContext *avctx, void *ptr_st,
867 GetBitContext *gb, float *out)
868 {
869 DecoderState *st = ptr_st;
870 float ol_gain = 0, ol_pitch_coef = 0, best_pitch_gain = 0, pitch_average = 0;
871 int m, pitch, wideband, ol_pitch = 0, best_pitch = 40;
872 SpeexContext *s = avctx->priv_data;
873 float innov[NB_SUBFRAME_SIZE];
874 float exc32[NB_SUBFRAME_SIZE];
875 float interp_qlsp[NB_ORDER];
876 float qlsp[NB_ORDER];
877 float ak[NB_ORDER];
878 float pitch_gain[3] = { 0 };
879
880 st->exc = st->exc_buf + 2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 6;
881
882 if (st->encode_submode) {
883 do { /* Search for next narrowband block (handle requests, skip wideband blocks) */
884 if (get_bits_left(gb) < 5)
885 return AVERROR_INVALIDDATA;
886 wideband = get_bits1(gb);
887 if (wideband) /* Skip wideband block (for compatibility) */ {
888 int submode, advance;
889
890 submode = get_bits(gb, SB_SUBMODE_BITS);
891 advance = wb_skip_table[submode];
892 advance -= SB_SUBMODE_BITS + 1;
893 if (advance < 0)
894 return AVERROR_INVALIDDATA;
895 skip_bits_long(gb, advance);
896
897 if (get_bits_left(gb) < 5)
898 return AVERROR_INVALIDDATA;
899 wideband = get_bits1(gb);
900 if (wideband) {
901 submode = get_bits(gb, SB_SUBMODE_BITS);
902 advance = wb_skip_table[submode];
903 advance -= SB_SUBMODE_BITS + 1;
904 if (advance < 0)
905 return AVERROR_INVALIDDATA;
906 skip_bits_long(gb, advance);
907 wideband = get_bits1(gb);
908 if (wideband) {
909 av_log(avctx, AV_LOG_ERROR, "more than two wideband layers found\n");
910 return AVERROR_INVALIDDATA;
911 }
912 }
913 }
914 if (get_bits_left(gb) < 4)
915 return AVERROR_INVALIDDATA;
916 m = get_bits(gb, 4);
917 if (m == 15) /* We found a terminator */ {
918 return AVERROR_INVALIDDATA;
919 } else if (m == 14) /* Speex in-band request */ {
920 int ret = speex_inband_handler(gb, st, &s->stereo);
921 if (ret)
922 return ret;
923 } else if (m == 13) /* User in-band request */ {
924 int ret = speex_default_user_handler(gb, st, NULL);
925 if (ret)
926 return ret;
927 } else if (m > 8) /* Invalid mode */ {
928 return AVERROR_INVALIDDATA;
929 }
930 } while (m > 8);
931
932 st->submodeID = m; /* Get the sub-mode that was used */
933 }
934
935 /* Shift all buffers by one frame */
936 memmove(st->exc_buf, st->exc_buf + NB_FRAME_SIZE, (2 * NB_PITCH_END + NB_SUBFRAME_SIZE + 12) * sizeof(float));
937
938 /* If null mode (no transmission), just set a couple things to zero */
939 if (st->submodes[st->submodeID] == NULL) {
940 float lpc[NB_ORDER];
941 float innov_gain = 0.f;
942
943 bw_lpc(0.93f, st->interp_qlpc, lpc, NB_ORDER);
944 innov_gain = compute_rms(st->exc, NB_FRAME_SIZE);
945 for (int i = 0; i < NB_FRAME_SIZE; i++)
946 st->exc[i] = speex_rand(innov_gain, &st->seed);
947
948 /* Final signal synthesis from excitation */
949 iir_mem(st->exc, lpc, out, NB_FRAME_SIZE, NB_ORDER, st->mem_sp);
950 st->count_lost = 0;
951
952 return 0;
953 }
954
955 /* Unquantize LSPs */
956 SUBMODE(lsp_unquant)(qlsp, NB_ORDER, gb);
957
958 /* Damp memory if a frame was lost and the LSP changed too much */
959 if (st->count_lost) {
960 float fact, lsp_dist = 0;
961
962 for (int i = 0; i < NB_ORDER; i++)
963 lsp_dist = lsp_dist + FFABS(st->old_qlsp[i] - qlsp[i]);
964 fact = .6f * exp(-.2f * lsp_dist);
965 for (int i = 0; i < NB_ORDER; i++)
966 st->mem_sp[i] = fact * st->mem_sp[i];
967 }
968
969 /* Handle first frame and lost-packet case */
970 if (st->first || st->count_lost)
971 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
972
973 /* Get open-loop pitch estimation for low bit-rate pitch coding */
974 if (SUBMODE(lbr_pitch) != -1)
975 ol_pitch = NB_PITCH_START + get_bits(gb, 7);
976
977 if (SUBMODE(forced_pitch_gain))
978 ol_pitch_coef = 0.066667f * get_bits(gb, 4);
979
980 /* Get global excitation gain */
981 ol_gain = expf(get_bits(gb, 5) / 3.5f);
982
983 if (st->submodeID == 1)
984 st->dtx_enabled = get_bits(gb, 4) == 15;
985
986 if (st->submodeID > 1)
987 st->dtx_enabled = 0;
988
989 for (int sub = 0; sub < NB_NB_SUBFRAMES; sub++) { /* Loop on subframes */
990 float *exc, *innov_save = NULL, tmp, ener;
991 int pit_min, pit_max, offset, q_energy;
992
993 offset = NB_SUBFRAME_SIZE * sub; /* Offset relative to start of frame */
994 exc = st->exc + offset; /* Excitation */
995 if (st->innov_save) /* Original signal */
996 innov_save = st->innov_save + offset;
997
998 SPEEX_MEMSET(exc, 0, NB_SUBFRAME_SIZE); /* Reset excitation */
999
1000 /* Adaptive codebook contribution */
1001 av_assert0(SUBMODE(ltp_unquant));
1002 /* Handle pitch constraints if any */
1003 if (SUBMODE(lbr_pitch) != -1) {
1004 int margin = SUBMODE(lbr_pitch);
1005
1006 if (margin) {
1007 pit_min = ol_pitch - margin + 1;
1008 pit_min = FFMAX(pit_min, NB_PITCH_START);
1009 pit_max = ol_pitch + margin;
1010 pit_max = FFMIN(pit_max, NB_PITCH_START);
1011 } else {
1012 pit_min = pit_max = ol_pitch;
1013 }
1014 } else {
1015 pit_min = NB_PITCH_START;
1016 pit_max = NB_PITCH_END;
1017 }
1018
1019 SUBMODE(ltp_unquant)(exc, exc32, pit_min, pit_max, ol_pitch_coef, SUBMODE(LtpParam),
1020 NB_SUBFRAME_SIZE, &pitch, pitch_gain, gb, st->count_lost, offset,
1021 st->last_pitch_gain, 0);
1022
1023 sanitize_values(exc32, -32000, 32000, NB_SUBFRAME_SIZE);
1024
1025 tmp = gain_3tap_to_1tap(pitch_gain);
1026
1027 pitch_average += tmp;
1028 if ((tmp > best_pitch_gain &&
1029 FFABS(2 * best_pitch - pitch) >= 3 &&
1030 FFABS(3 * best_pitch - pitch) >= 4 &&
1031 FFABS(4 * best_pitch - pitch) >= 5) ||
1032 (tmp > .6f * best_pitch_gain &&
1033 (FFABS(best_pitch - 2 * pitch) < 3 ||
1034 FFABS(best_pitch - 3 * pitch) < 4 ||
1035 FFABS(best_pitch - 4 * pitch) < 5)) ||
1036 ((.67f * tmp) > best_pitch_gain &&
1037 (FFABS(2 * best_pitch - pitch) < 3 ||
1038 FFABS(3 * best_pitch - pitch) < 4 ||
1039 FFABS(4 * best_pitch - pitch) < 5))) {
1040 best_pitch = pitch;
1041 if (tmp > best_pitch_gain)
1042 best_pitch_gain = tmp;
1043 }
1044
1045 memset(innov, 0, sizeof(innov));
1046
1047 /* Decode sub-frame gain correction */
1048 if (SUBMODE(have_subframe_gain) == 3) {
1049 q_energy = get_bits(gb, 3);
1050 ener = exc_gain_quant_scal3[q_energy] * ol_gain;
1051 } else if (SUBMODE(have_subframe_gain) == 1) {
1052 q_energy = get_bits1(gb);
1053 ener = exc_gain_quant_scal1[q_energy] * ol_gain;
1054 } else {
1055 ener = ol_gain;
1056 }
1057
1058 av_assert0(SUBMODE(innovation_unquant));
1059 /* Fixed codebook contribution */
1060 SUBMODE(innovation_unquant)(innov, SUBMODE(innovation_params), NB_SUBFRAME_SIZE, gb, &st->seed);
1061 /* De-normalize innovation and update excitation */
1062
1063 signal_mul(innov, innov, ener, NB_SUBFRAME_SIZE);
1064
1065 /* Decode second codebook (only for some modes) */
1066 if (SUBMODE(double_codebook)) {
1067 float innov2[NB_SUBFRAME_SIZE] = { 0 };
1068
1069 SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), NB_SUBFRAME_SIZE, gb, &st->seed);
1070 signal_mul(innov2, innov2, 0.454545f * ener, NB_SUBFRAME_SIZE);
1071 for (int i = 0; i < NB_SUBFRAME_SIZE; i++)
1072 innov[i] += innov2[i];
1073 }
1074 for (int i = 0; i < NB_SUBFRAME_SIZE; i++)
1075 exc[i] = exc32[i] + innov[i];
1076 if (innov_save)
1077 memcpy(innov_save, innov, sizeof(innov));
1078
1079 /* Vocoder mode */
1080 if (st->submodeID == 1) {
1081 float g = ol_pitch_coef;
1082
1083 g = av_clipf(1.5f * (g - .2f), 0.f, 1.f);
1084
1085 SPEEX_MEMSET(exc, 0, NB_SUBFRAME_SIZE);
1086 while (st->voc_offset < NB_SUBFRAME_SIZE) {
1087 if (st->voc_offset >= 0)
1088 exc[st->voc_offset] = sqrtf(2.f * ol_pitch) * (g * ol_gain);
1089 st->voc_offset += ol_pitch;
1090 }
1091 st->voc_offset -= NB_SUBFRAME_SIZE;
1092
1093 for (int i = 0; i < NB_SUBFRAME_SIZE; i++) {
1094 float exci = exc[i];
1095 exc[i] = (.7f * exc[i] + .3f * st->voc_m1) + ((1.f - .85f * g) * innov[i]) - .15f * g * st->voc_m2;
1096 st->voc_m1 = exci;
1097 st->voc_m2 = innov[i];
1098 st->voc_mean = .8f * st->voc_mean + .2f * exc[i];
1099 exc[i] -= st->voc_mean;
1100 }
1101 }
1102 }
1103
1104 if (st->lpc_enh_enabled && SUBMODE(comb_gain) > 0 && !st->count_lost) {
1105 multicomb(st->exc - NB_SUBFRAME_SIZE, out, st->interp_qlpc, NB_ORDER,
1106 2 * NB_SUBFRAME_SIZE, best_pitch, 40, SUBMODE(comb_gain));
1107 multicomb(st->exc + NB_SUBFRAME_SIZE, out + 2 * NB_SUBFRAME_SIZE,
1108 st->interp_qlpc, NB_ORDER, 2 * NB_SUBFRAME_SIZE, best_pitch, 40,
1109 SUBMODE(comb_gain));
1110 } else {
1111 SPEEX_COPY(out, &st->exc[-NB_SUBFRAME_SIZE], NB_FRAME_SIZE);
1112 }
1113
1114 /* If the last packet was lost, re-scale the excitation to obtain the same
1115 * energy as encoded in ol_gain */
1116 if (st->count_lost) {
1117 float exc_ener, gain;
1118
1119 exc_ener = compute_rms(st->exc, NB_FRAME_SIZE);
1120 av_assert0(exc_ener + 1.f > 0.f);
1121 gain = fminf(ol_gain / (exc_ener + 1.f), 2.f);
1122 for (int i = 0; i < NB_FRAME_SIZE; i++) {
1123 st->exc[i] *= gain;
1124 out[i] = st->exc[i - NB_SUBFRAME_SIZE];
1125 }
1126 }
1127
1128 for (int sub = 0; sub < NB_NB_SUBFRAMES; sub++) { /* Loop on subframes */
1129 const int offset = NB_SUBFRAME_SIZE * sub; /* Offset relative to start of frame */
1130 float pi_g = 1.f, *sp = out + offset; /* Original signal */
1131
1132 lsp_interpolate(st->old_qlsp, qlsp, interp_qlsp, NB_ORDER, sub, NB_NB_SUBFRAMES, 0.002f);
1133 lsp_to_lpc(interp_qlsp, ak, NB_ORDER); /* Compute interpolated LPCs (unquantized) */
1134
1135 for (int i = 0; i < NB_ORDER; i += 2) /* Compute analysis filter at w=pi */
1136 pi_g += ak[i + 1] - ak[i];
1137 st->pi_gain[sub] = pi_g;
1138 st->exc_rms[sub] = compute_rms(st->exc + offset, NB_SUBFRAME_SIZE);
1139
1140 iir_mem(sp, st->interp_qlpc, sp, NB_SUBFRAME_SIZE, NB_ORDER, st->mem_sp);
1141
1142 memcpy(st->interp_qlpc, ak, sizeof(st->interp_qlpc));
1143 }
1144
1145 if (st->highpass_enabled)
1146 highpass(out, out, NB_FRAME_SIZE, st->mem_hp, st->is_wideband);
1147
1148 /* Store the LSPs for interpolation in the next frame */
1149 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
1150
1151 st->count_lost = 0;
1152 st->last_pitch = best_pitch;
1153 st->last_pitch_gain = .25f * pitch_average;
1154 st->last_ol_gain = ol_gain;
1155 st->first = 0;
1156
1157 return 0;
1158 }
1159
1160 static void qmf_synth(const float *x1, const float *x2, const float *a, float *y, int N, int M, float *mem1, float *mem2)
1161 {
1162 const int M2 = M >> 1, N2 = N >> 1;
1163 float xx1[352], xx2[352];
1164
1165 for (int i = 0; i < N2; i++)
1166 xx1[i] = x1[N2-1-i];
1167 for (int i = 0; i < M2; i++)
1168 xx1[N2+i] = mem1[2*i+1];
1169 for (int i = 0; i < N2; i++)
1170 xx2[i] = x2[N2-1-i];
1171 for (int i = 0; i < M2; i++)
1172 xx2[N2+i] = mem2[2*i+1];
1173
1174 for (int i = 0; i < N2; i += 2) {
1175 float y0, y1, y2, y3;
1176 float x10, x20;
1177
1178 y0 = y1 = y2 = y3 = 0.f;
1179 x10 = xx1[N2-2-i];
1180 x20 = xx2[N2-2-i];
1181
1182 for (int j = 0; j < M2; j += 2) {
1183 float x11, x21;
1184 float a0, a1;
1185
1186 a0 = a[2*j];
1187 a1 = a[2*j+1];
1188 x11 = xx1[N2-1+j-i];
1189 x21 = xx2[N2-1+j-i];
1190
1191 y0 += a0 * (x11-x21);
1192 y1 += a1 * (x11+x21);
1193 y2 += a0 * (x10-x20);
1194 y3 += a1 * (x10+x20);
1195 a0 = a[2*j+2];
1196 a1 = a[2*j+3];
1197 x10 = xx1[N2+j-i];
1198 x20 = xx2[N2+j-i];
1199
1200 y0 += a0 * (x10-x20);
1201 y1 += a1 * (x10+x20);
1202 y2 += a0 * (x11-x21);
1203 y3 += a1 * (x11+x21);
1204 }
1205 y[2 * i ] = 2.f * y0;
1206 y[2 * i+1] = 2.f * y1;
1207 y[2 * i+2] = 2.f * y2;
1208 y[2 * i+3] = 2.f * y3;
1209 }
1210
1211 for (int i = 0; i < M2; i++)
1212 mem1[2*i+1] = xx1[i];
1213 for (int i = 0; i < M2; i++)
1214 mem2[2*i+1] = xx2[i];
1215 }
1216
1217 static int sb_decode(AVCodecContext *avctx, void *ptr_st,
1218 GetBitContext *gb, float *out)
1219 {
1220 SpeexContext *s = avctx->priv_data;
1221 DecoderState *st = ptr_st;
1222 float low_pi_gain[NB_NB_SUBFRAMES];
1223 float low_exc_rms[NB_NB_SUBFRAMES];
1224 float interp_qlsp[NB_ORDER];
1225 int ret, wideband;
1226 float *low_innov_alias;
1227 float qlsp[NB_ORDER];
1228 float ak[NB_ORDER];
1229 const SpeexMode *mode;
1230
1231 mode = st->mode;
1232
1233 if (st->modeID > 0) {
1234 low_innov_alias = out + st->frame_size;
1235 s->st[st->modeID - 1].innov_save = low_innov_alias;
1236 ret = speex_modes[st->modeID - 1].decode(avctx, &s->st[st->modeID - 1], gb, out);
1237 if (ret < 0)
1238 return ret;
1239 }
1240
1241 if (st->encode_submode) { /* Check "wideband bit" */
1242 if (get_bits_left(gb) > 0)
1243 wideband = show_bits1(gb);
1244 else
1245 wideband = 0;
1246 if (wideband) { /* Regular wideband frame, read the submode */
1247 wideband = get_bits1(gb);
1248 st->submodeID = get_bits(gb, SB_SUBMODE_BITS);
1249 } else { /* Was a narrowband frame, set "null submode" */
1250 st->submodeID = 0;
1251 }
1252 if (st->submodeID != 0 && st->submodes[st->submodeID] == NULL)
1253 return AVERROR_INVALIDDATA;
1254 }
1255
1256 /* If null mode (no transmission), just set a couple things to zero */
1257 if (st->submodes[st->submodeID] == NULL) {
1258 for (int i = 0; i < st->frame_size; i++)
1259 out[st->frame_size + i] = 1e-15f;
1260
1261 st->first = 1;
1262
1263 /* Final signal synthesis from excitation */
1264 iir_mem(out + st->frame_size, st->interp_qlpc, out + st->frame_size, st->frame_size, st->lpc_size, st->mem_sp);
1265
1266 qmf_synth(out, out + st->frame_size, h0, out, st->full_frame_size, QMF_ORDER, st->g0_mem, st->g1_mem);
1267
1268 return 0;
1269 }
1270
1271 memcpy(low_pi_gain, s->st[st->modeID - 1].pi_gain, sizeof(low_pi_gain));
1272 memcpy(low_exc_rms, s->st[st->modeID - 1].exc_rms, sizeof(low_exc_rms));
1273
1274 SUBMODE(lsp_unquant)(qlsp, st->lpc_size, gb);
1275
1276 if (st->first)
1277 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
1278
1279 for (int sub = 0; sub < st->nb_subframes; sub++) {
1280 float filter_ratio, el, rl, rh;
1281 float *innov_save = NULL, *sp;
1282 float exc[80];
1283 int offset;
1284
1285 offset = st->subframe_size * sub;
1286 sp = out + st->frame_size + offset;
1287 /* Pointer for saving innovation */
1288 if (st->innov_save) {
1289 innov_save = st->innov_save + 2 * offset;
1290 SPEEX_MEMSET(innov_save, 0, 2 * st->subframe_size);
1291 }
1292
1293 av_assert0(st->nb_subframes > 0);
1294 lsp_interpolate(st->old_qlsp, qlsp, interp_qlsp, st->lpc_size, sub, st->nb_subframes, 0.05f);
1295 lsp_to_lpc(interp_qlsp, ak, st->lpc_size);
1296
1297 /* Calculate reponse ratio between the low and high filter in the middle
1298 of the band (4000 Hz) */
1299 st->pi_gain[sub] = 1.f;
1300 rh = 1.f;
1301 for (int i = 0; i < st->lpc_size; i += 2) {
1302 rh += ak[i + 1] - ak[i];
1303 st->pi_gain[sub] += ak[i] + ak[i + 1];
1304 }
1305
1306 rl = low_pi_gain[sub];
1307 filter_ratio = (rl + .01f) / (rh + .01f);
1308
1309 SPEEX_MEMSET(exc, 0, st->subframe_size);
1310 if (!SUBMODE(innovation_unquant)) {
1311 const int x = get_bits(gb, 5);
1312 const float g = expf(.125f * (x - 10)) / filter_ratio;
1313
1314 for (int i = 0; i < st->subframe_size; i += 2) {
1315 exc[i ] = mode->folding_gain * low_innov_alias[offset + i ] * g;
1316 exc[i + 1] = -mode->folding_gain * low_innov_alias[offset + i + 1] * g;
1317 }
1318 } else {
1319 float gc, scale;
1320
1321 el = low_exc_rms[sub];
1322 gc = 0.87360f * gc_quant_bound[get_bits(gb, 4)];
1323
1324 if (st->subframe_size == 80)
1325 gc *= M_SQRT2;
1326
1327 scale = (gc * el) / filter_ratio;
1328 SUBMODE(innovation_unquant)
1329 (exc, SUBMODE(innovation_params), st->subframe_size,
1330 gb, &st->seed);
1331
1332 signal_mul(exc, exc, scale, st->subframe_size);
1333 if (SUBMODE(double_codebook)) {
1334 float innov2[80];
1335
1336 SPEEX_MEMSET(innov2, 0, st->subframe_size);
1337 SUBMODE(innovation_unquant)(innov2, SUBMODE(innovation_params), st->subframe_size, gb, &st->seed);
1338 signal_mul(innov2, innov2, 0.4f * scale, st->subframe_size);
1339 for (int i = 0; i < st->subframe_size; i++)
1340 exc[i] += innov2[i];
1341 }
1342 }
1343
1344 if (st->innov_save) {
1345 for (int i = 0; i < st->subframe_size; i++)
1346 innov_save[2 * i] = exc[i];
1347 }
1348
1349 iir_mem(st->exc_buf, st->interp_qlpc, sp, st->subframe_size, st->lpc_size, st->mem_sp);
1350 memcpy(st->exc_buf, exc, sizeof(exc));
1351 memcpy(st->interp_qlpc, ak, sizeof(st->interp_qlpc));
1352 st->exc_rms[sub] = compute_rms(st->exc_buf, st->subframe_size);
1353 }
1354
1355 qmf_synth(out, out + st->frame_size, h0, out, st->full_frame_size, QMF_ORDER, st->g0_mem, st->g1_mem);
1356 memcpy(st->old_qlsp, qlsp, sizeof(st->old_qlsp));
1357
1358 st->first = 0;
1359
1360 return 0;
1361 }
1362
1363 static int decoder_init(SpeexContext *s, DecoderState *st, const SpeexMode *mode)
1364 {
1365 st->mode = mode;
1366 st->modeID = mode->modeID;
1367
1368 st->first = 1;
1369 st->encode_submode = 1;
1370 st->is_wideband = st->modeID > 0;
1371 st->innov_save = NULL;
1372
1373 st->submodes = mode->submodes;
1374 st->submodeID = mode->default_submode;
1375 st->subframe_size = mode->subframe_size;
1376 st->lpc_size = mode->lpc_size;
1377 st->full_frame_size = (1 + (st->modeID > 0)) * mode->frame_size;
1378 st->nb_subframes = mode->frame_size / mode->subframe_size;
1379 st->frame_size = mode->frame_size;
1380
1381 st->lpc_enh_enabled = 1;
1382
1383 st->last_pitch = 40;
1384 st->count_lost = 0;
1385 st->seed = 1000;
1386 st->last_ol_gain = 0;
1387
1388 st->voc_m1 = st->voc_m2 = st->voc_mean = 0;
1389 st->voc_offset = 0;
1390 st->dtx_enabled = 0;
1391 st->highpass_enabled = mode->modeID == 0;
1392
1393 return 0;
1394 }
1395
1396 6 static int parse_speex_extradata(AVCodecContext *avctx,
1397 const uint8_t *extradata, int extradata_size)
1398 {
1399 6 SpeexContext *s = avctx->priv_data;
1400 6 const uint8_t *buf = extradata;
1401
1402
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6 if (memcmp(buf, "Speex ", 8))
1403 6 return AVERROR_INVALIDDATA;
1404
1405 buf += 28;
1406
1407 s->version_id = bytestream_get_le32(&buf);
1408 buf += 4;
1409 s->rate = bytestream_get_le32(&buf);
1410 if (s->rate <= 0)
1411 return AVERROR_INVALIDDATA;
1412 s->mode = bytestream_get_le32(&buf);
1413 if (s->mode < 0 || s->mode >= SPEEX_NB_MODES)
1414 return AVERROR_INVALIDDATA;
1415 s->bitstream_version = bytestream_get_le32(&buf);
1416 if (s->bitstream_version != 4)
1417 return AVERROR_INVALIDDATA;
1418 s->nb_channels = bytestream_get_le32(&buf);
1419 if (s->nb_channels <= 0 || s->nb_channels > 2)
1420 return AVERROR_INVALIDDATA;
1421 s->bitrate = bytestream_get_le32(&buf);
1422 s->frame_size = bytestream_get_le32(&buf);
1423 if (s->frame_size < NB_FRAME_SIZE << s->mode)
1424 return AVERROR_INVALIDDATA;
1425 s->vbr = bytestream_get_le32(&buf);
1426 s->frames_per_packet = bytestream_get_le32(&buf);
1427 if (s->frames_per_packet <= 0 ||
1428 s->frames_per_packet > 64 ||
1429 s->frames_per_packet >= INT32_MAX / s->nb_channels / s->frame_size)
1430 return AVERROR_INVALIDDATA;
1431 s->extra_headers = bytestream_get_le32(&buf);
1432
1433 return 0;
1434 }
1435
1436 6 static av_cold int speex_decode_init(AVCodecContext *avctx)
1437 {
1438 6 SpeexContext *s = avctx->priv_data;
1439 int ret;
1440
1441 6 s->fdsp = avpriv_float_dsp_alloc(0);
1442
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6 if (!s->fdsp)
1443 return AVERROR(ENOMEM);
1444
1445
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6 if (avctx->extradata && avctx->extradata_size >= 80) {
1446 6 ret = parse_speex_extradata(avctx, avctx->extradata, avctx->extradata_size);
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6 if (ret < 0)
1448 6 return ret;
1449 } else {
1450 s->rate = avctx->sample_rate;
1451 if (s->rate <= 0)
1452 return AVERROR_INVALIDDATA;
1453
1454 s->nb_channels = avctx->ch_layout.nb_channels;
1455 if (s->nb_channels <= 0)
1456 return AVERROR_INVALIDDATA;
1457
1458 switch (s->rate) {
1459 case 8000: s->mode = 0; break;
1460 case 16000: s->mode = 1; break;
1461 case 32000: s->mode = 2; break;
1462 default: s->mode = 2;
1463 }
1464
1465 s->frames_per_packet = 64;
1466 s->frame_size = NB_FRAME_SIZE << s->mode;
1467 }
1468
1469 if (avctx->codec_tag == MKTAG('S', 'P', 'X', 'N')) {
1470 int quality;
1471
1472 if (!avctx->extradata || avctx->extradata && avctx->extradata_size < 47) {
1473 av_log(avctx, AV_LOG_ERROR, "Missing or invalid extradata.\n");
1474 return AVERROR_INVALIDDATA;
1475 }
1476
1477 quality = avctx->extradata[37];
1478 if (quality > 10) {
1479 av_log(avctx, AV_LOG_ERROR, "Unsupported quality mode %d.\n", quality);
1480 return AVERROR_PATCHWELCOME;
1481 }
1482
1483 s->pkt_size = ((const uint8_t[]){ 5, 10, 15, 20, 20, 28, 28, 38, 38, 46, 62 })[quality];
1484
1485 s->mode = 0;
1486 s->nb_channels = 1;
1487 s->rate = avctx->sample_rate;
1488 if (s->rate <= 0)
1489 return AVERROR_INVALIDDATA;
1490 s->frames_per_packet = 1;
1491 s->frame_size = NB_FRAME_SIZE;
1492 }
1493
1494 if (s->bitrate > 0)
1495 avctx->bit_rate = s->bitrate;
1496 av_channel_layout_uninit(&avctx->ch_layout);
1497 avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
1498 avctx->ch_layout.nb_channels = s->nb_channels;
1499 avctx->sample_rate = s->rate;
1500 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
1501
1502 for (int m = 0; m <= s->mode; m++) {
1503 ret = decoder_init(s, &s->st[m], &speex_modes[m]);
1504 if (ret < 0)
1505 return ret;
1506 }
1507
1508 s->stereo.balance = 1.f;
1509 s->stereo.e_ratio = .5f;
1510 s->stereo.smooth_left = 1.f;
1511 s->stereo.smooth_right = 1.f;
1512
1513 return 0;
1514 }
1515
1516 static void speex_decode_stereo(float *data, int frame_size, StereoState *stereo)
1517 {
1518 float balance, e_left, e_right, e_ratio;
1519
1520 balance = stereo->balance;
1521 e_ratio = stereo->e_ratio;
1522
1523 /* These two are Q14, with max value just below 2. */
1524 e_right = 1.f / sqrtf(e_ratio * (1.f + balance));
1525 e_left = sqrtf(balance) * e_right;
1526
1527 for (int i = frame_size - 1; i >= 0; i--) {
1528 float tmp = data[i];
1529 stereo->smooth_left = stereo->smooth_left * 0.98f + e_left * 0.02f;
1530 stereo->smooth_right = stereo->smooth_right * 0.98f + e_right * 0.02f;
1531 data[2 * i ] = stereo->smooth_left * tmp;
1532 data[2 * i + 1] = stereo->smooth_right * tmp;
1533 }
1534 }
1535
1536 static int speex_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1537 int *got_frame_ptr, AVPacket *avpkt)
1538 {
1539 SpeexContext *s = avctx->priv_data;
1540 int frames_per_packet = s->frames_per_packet;
1541 const float scale = 1.f / 32768.f;
1542 int buf_size = avpkt->size;
1543 float *dst;
1544 int ret;
1545
1546 if (s->pkt_size && avpkt->size == 62)
1547 buf_size = s->pkt_size;
1548 if ((ret = init_get_bits8(&s->gb, avpkt->data, buf_size)) < 0)
1549 return ret;
1550
1551 frame->nb_samples = FFALIGN(s->frame_size * frames_per_packet, 4);
1552 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1553 return ret;
1554
1555 dst = (float *)frame->extended_data[0];
1556 for (int i = 0; i < frames_per_packet; i++) {
1557 ret = speex_modes[s->mode].decode(avctx, &s->st[s->mode], &s->gb, dst + i * s->frame_size);
1558 if (ret < 0)
1559 return ret;
1560 if (avctx->ch_layout.nb_channels == 2)
1561 speex_decode_stereo(dst + i * s->frame_size, s->frame_size, &s->stereo);
1562 if (get_bits_left(&s->gb) < 5 ||
1563 show_bits(&s->gb, 5) == 15) {
1564 frames_per_packet = i + 1;
1565 break;
1566 }
1567 }
1568
1569 dst = (float *)frame->extended_data[0];
1570 s->fdsp->vector_fmul_scalar(dst, dst, scale, frame->nb_samples * frame->ch_layout.nb_channels);
1571 frame->nb_samples = s->frame_size * frames_per_packet;
1572
1573 *got_frame_ptr = 1;
1574
1575 return (get_bits_count(&s->gb) + 7) >> 3;
1576 }
1577
1578 6 static av_cold int speex_decode_close(AVCodecContext *avctx)
1579 {
1580 6 SpeexContext *s = avctx->priv_data;
1581 6 av_freep(&s->fdsp);
1582 6 return 0;
1583 }
1584
1585 const FFCodec ff_speex_decoder = {
1586 .p.name = "speex",
1587 CODEC_LONG_NAME("Speex"),
1588 .p.type = AVMEDIA_TYPE_AUDIO,
1589 .p.id = AV_CODEC_ID_SPEEX,
1590 .init = speex_decode_init,
1591 FF_CODEC_DECODE_CB(speex_decode_frame),
1592 .close = speex_decode_close,
1593 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1594 .priv_data_size = sizeof(SpeexContext),
1595 .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1596 };
1597